VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
rajil.s at gmail.com Guest
|
Posted: Sat Feb 13, 2016 9:55 pm Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
Hello,
I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.
Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?
The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM
---------------------------siptrace--------------------------------
recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0
------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
------------------------------------------------------------------------
ACK sip:303@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0
------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]
Thanks
Rajil
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org |
|
Back to top |
|
|
gmaruzz at gmail.com Guest
|
Posted: Sun Feb 14, 2016 12:30 pm Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
How you originate the call? Is a bridge? From which phone?
Also, please pastebin the complete sip trace (from start of leg A to end of both legs) and put here a link to pastebin
Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> ha scritto: Quote: | Hello,
I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5 (303@192.168.1.5)
the call works fine.
Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?
The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM
---------------------------siptrace--------------------------------
recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0
------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
------------------------------------------------------------------------
ACK sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email]) SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0
------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 (303@192.168.1.5) entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 (303@192.168.1.5) [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]
Thanks
Rajil
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
|
|
Back to top |
|
|
rajil.s at gmail.com Guest
|
Posted: Sun Feb 14, 2016 2:06 pm Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
The siptrace is at http://pastebin.com/xiGqtj1Y
The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.
On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
Quote: | How you originate the call? Is a bridge? From which phone?
Also, please pastebin the complete sip trace (from start of leg A to end of
both legs) and put here a link to pastebin
Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com> ha scritto:
Quote: |
Hello,
I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.
Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?
The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM
---------------------------siptrace--------------------------------
recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0
------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
------------------------------------------------------------------------
ACK sip:303@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0
------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]
Thanks
Rajil
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org |
|
Back to top |
|
|
rtreleaven at bunnykic... Guest
|
Posted: Sun Feb 14, 2016 2:35 pm Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
Look at the invite starting at 480. You are only offering opus to the callee. On Feb 14, 2016 2:05 PM, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> wrote: |
|
Back to top |
|
|
rtreleaven at bunnykic... Guest
|
|
Back to top |
|
|
rajil.s at gmail.com Guest
|
Posted: Sun Feb 14, 2016 4:39 pm Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically. I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?
On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
Quote: | fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Quote: |
The siptrace is at http://pastebin.com/xiGqtj1Y
The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.
On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com>
wrote:
Quote: | How you originate the call? Is a bridge? From which phone?
Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin
Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com> ha scritto:
Quote: |
Hello,
I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.
Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?
The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM
---------------------------siptrace--------------------------------
recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0
------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
------------------------------------------------------------------------
ACK sip:303@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0
------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]
Thanks
Rajil
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org |
|
Back to top |
|
|
krice at freeswitch.org Guest
|
Posted: Sun Feb 14, 2016 5:08 pm Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it
Sent from my iPhone
Quote: | On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically. I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?
Quote: | On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
Quote: | On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
The siptrace is at http://pastebin.com/xiGqtj1Y
The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.
On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com>
wrote:
Quote: | How you originate the call? Is a bridge? From which phone?
Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin
Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com> ha scritto:
Quote: |
Hello,
I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.
Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?
The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM
---------------------------siptrace--------------------------------
recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0
------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
------------------------------------------------------------------------
ACK sip:303@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0
------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]
Thanks
Rajil
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org |
|
Back to top |
|
|
colton.conor at gmail.com Guest
|
Posted: Sun Feb 14, 2016 7:22 pm Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.
Plus if you use TLS for encryption and security then you are already using TCP right?
On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice@freeswitch.org (krice@freeswitch.org)> wrote:
|
|
Back to top |
|
|
mike at jerris.com Guest
|
Posted: Mon Feb 15, 2016 10:35 am Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
any device that even remotely follows sipspecs supports TCP. Most phones I have seen do.
On Sunday, February 14, 2016, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
Quote: | So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.
Plus if you use TLS for encryption and security then you are already using TCP right?
On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <[url=javascript:_e(%7B%7D,'cvml','krice@freeswitch.org');]krice@freeswitch.org[/url]> wrote:
Quote: | This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it
Sent from my iPhone
Quote: | On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <[url=javascript:_e(%7B%7D,'cvml','rajil.s@gmail.com');]rajil.s@gmail.com[/url]> wrote:
Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically. I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?
Quote: | On 14 February 2016 at 13:56, Russell Treleaven <[url=javascript:_e(%7B%7D,'cvml','rtreleaven@bunnykick.ca');]rtreleaven@bunnykick.ca[/url]> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
Quote: | On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <[url=javascript:_e(%7B%7D,'cvml','rajil.s@gmail.com');]rajil.s@gmail.com[/url]> wrote:
The siptrace is at http://pastebin.com/xiGqtj1Y
The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.
On 14 February 2016 at 11:28, Giovanni Maruzzelli <[url=javascript:_e(%7B%7D,'cvml','gmaruzz@gmail.com');]gmaruzz@gmail.com[/url]>
wrote:
Quote: | How you originate the call? Is a bridge? From which phone?
Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin
Il 14/Feb/2016 03:54, "Rajil Saraswat" <[url=javascript:_e(%7B%7D,'cvml','rajil.s@gmail.com');]rajil.s@gmail.com[/url]> ha scritto:
Quote: |
Hello,
I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/[url=javascript:_e(%7B%7D,'cvml','303@192.168.1.5');]303@192.168.1.5[/url]
the call works fine.
Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?
The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM
---------------------------siptrace--------------------------------
recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <[url=javascript:_e(%7B%7D,'cvml','sip:202@192.168.1.111');]sip:202@192.168.1.111[/url]>;tag=DFX0FUvr2vNcm
To: <[url=javascript:_e(%7B%7D,'cvml','sip:303@192.168.1.5');]sip:303@192.168.1.5[/url]>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0
------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
------------------------------------------------------------------------
ACK [url=javascript:_e(%7B%7D,'cvml','sip:303@192.168.1.5');]sip:303@192.168.1.5[/url] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <[url=javascript:_e(%7B%7D,'cvml','sip:202@192.168.1.111');]sip:202@192.168.1.111[/url]>;tag=DFX0FUvr2vNcm
To: <[url=javascript:_e(%7B%7D,'cvml','sip:303@192.168.1.5');]sip:303@192.168.1.5[/url]>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0
------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/[url=javascript:_e(%7B%7D,'cvml','303@192.168.1.5');]303@192.168.1.5[/url] entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/[url=javascript:_e(%7B%7D,'cvml','303@192.168.1.5');]303@192.168.1.5[/url] [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]
Thanks
Rajil
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[url=javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');]consulting@freeswitch.org[/url]
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
[url=javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');]FreeSWITCH-users@lists.freeswitch.org[/url]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[url=javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');]consulting@freeswitch.org[/url]
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
[url=javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');]FreeSWITCH-users@lists.freeswitch.org[/url]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[url=javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');]consulting@freeswitch.org[/url]
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
[url=javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');]FreeSWITCH-users@lists.freeswitch.org[/url]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[url=javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');]consulting@freeswitch.org[/url]
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
[url=javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');]FreeSWITCH-users@lists.freeswitch.org[/url]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[url=javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');]consulting@freeswitch.org[/url]
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
[url=javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');]FreeSWITCH-users@lists.freeswitch.org[/url]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[url=javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');]consulting@freeswitch.org[/url]
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
[url=javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');]FreeSWITCH-users@lists.freeswitch.org[/url]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
|
|
|
Back to top |
|
|
colton.conor at gmail.com Guest
|
Posted: Mon Feb 15, 2016 11:06 am Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior.
On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote:
Quote: | any device that even remotely follows sipspecs supports TCP. Most phones I have seen do.
On Sunday, February 14, 2016, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
Quote: | So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.
Plus if you use TLS for encryption and security then you are already using TCP right?
On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice@freeswitch.org> wrote:
Quote: | This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it
Sent from my iPhone
Quote: | On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically. I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?
Quote: | On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
Quote: | On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
The siptrace is at http://pastebin.com/xiGqtj1Y
The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.
On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com>
wrote:
Quote: | How you originate the call? Is a bridge? From which phone?
Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin
Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com> ha scritto:
Quote: |
Hello,
I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.
Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?
The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM
---------------------------siptrace--------------------------------
recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0
------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
------------------------------------------------------------------------
ACK sip:303@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0
------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]
Thanks
Rajil
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
|
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
|
|
Back to top |
|
|
mike at jerris.com Guest
|
Posted: Mon Feb 15, 2016 11:10 am Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
It is heavier but I think that otherwise is superior. Quote: | On Feb 15, 2016, at 11:04 AM, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior.
On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote: Quote: | any device that even remotely follows sipspecs supports TCP. Most phones I have seen do.On Sunday, February 14, 2016, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
|
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org (consulting@freeswitch.org) FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org |
_________________________________________________________________________Professional FreeSWITCH Consulting Services: consulting@freeswitch.org (consulting@freeswitch.org)http://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
[/quote] |
|
Back to top |
|
|
krice at freeswitch.org Guest
|
Posted: Mon Feb 15, 2016 11:27 am Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
The problem isn’t necessarily the devices, but there is also the carriers…
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Colton Conor
Sent: Monday, February 15, 2016 10:04 AM
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior.
On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote: |
|
Back to top |
|
|
colton.conor at gmail.com Guest
|
Posted: Mon Feb 15, 2016 11:31 am Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
True,
But freeswitch talking to the carriers is almost always UPD.
However, freeswitch talking to the clients I would say TCP would be idea. So its almost like freeswitch is trancoding from TCP to UDP too
On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice@freeswitch.org (krice@freeswitch.org)> wrote:
|
|
Back to top |
|
|
krice at freeswitch.org Guest
|
Posted: Mon Feb 15, 2016 11:37 am Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
The problem still exists for expanding SDPs… using TCP to the user/device then trying to send the same thing out to the carrier over UDP is what was causing the problem in the first place… so the decision was made to prevent those problems we’ll only offer what the device offers and not expand the number of codecs even further increasing the already bloated SDPs to the point where they fragment over UDP and get dropped…
So is TCP better for some things, yes it is, however, the lack of market wide support for it with carriers makes it a pain in the ass even tho the RFCs specifically say you MUST support both UDP and TCP for SIP, but certain VoIP softwares out there only implemented UDP many years ago and now we’re stuck with that legacy
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Colton Conor
Sent: Monday, February 15, 2016 10:30 AM
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
True,
But freeswitch talking to the carriers is almost always UPD.
However, freeswitch talking to the clients I would say TCP would be idea. So its almost like freeswitch is trancoding from TCP to UDP too
On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice@freeswitch.org (krice@freeswitch.org)> wrote: |
|
Back to top |
|
|
bobjectsfreeswitch at ... Guest
|
Posted: Mon Feb 15, 2016 1:28 pm Post subject: [Freeswitch-users] Freeswitch doesnt transcode |
|
|
On Sun, Feb 14, 2016 at 1:56 PM, Russell Treleaven <rtreleaven@bunnykick.ca (rtreleaven@bunnykick.ca)> wrote:
That is a really lucid and complete description of CODEC negotiation in FS. Thanks for the link, and thanks to the original author for writing it.
Bob |
|
Back to top |
|
|
|
|
|
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|